0 Replies Latest reply on Aug 14, 2017 3:12 PM by sroberts

    Adding a 2nd PBX

    sroberts New Member

      Hope someone can provide a good learning resource for this, or pick up a few things I am doing wrong.  I'm trying to setup FreePBX to test, but we are not replacing the current PBX.  Disclaimer that my FreePBX may be all wrong, but judging by my debugging, the Adtran 908e is not right yet either.

       

      We have a SIP trunk for local numbers and outgoing, and another SIP trunk for toll free.

       

      I added voice trunk T04 for the new test PBX, changed the accept cost of a group of DID's, and created voice grouped-trunk SIP_GROUP_2 with the new trunk and lower accept cost.  The trunks and grouped-trunks are as follows:

       

      voice trunk T01 type sip

        description "SIP TRUNK"

        caller-id-override number-inbound 9 if-no-cpn

        sip-server primary 192.168.1.10

        sip-server secondary 192.168.1.20

        check-supported replaces

      !

      voice trunk T02 type sip

        description "Toll Free"

        sip-server primary 67.x.x.x

        authentication username "asdfg" password encrypted "asdfg"

      !

      voice trunk T03 type sip

        description "Local DID"

        sip-server primary 8.x.x.x

      !

      voice trunk T04 type sip

        description "Test PBX"

        sip-server primary 192.168.1.8

        grammar from host local

        transfer-mode network

      !

      voice grouped-trunk "SIP GROUP"

        description "Production Servers"

        trunk T01

        accept $ cost 0

        accept 555999640X cost 0

        accept 555999641X cost 0

        accept 555999642X cost 0

        accept 555999643X cost 5

      !

      voice grouped-trunk LEVEL3_SIP

        description "TOLL FREE"

        trunk T02

        accept 5554069600 cost 0

        accept 18001234567 cost 0

        accept $ cost 10

      !

      voice grouped-trunk SIP_LOCAL

        trunk T03

        accept $ cost 5

        accept 411 cost 0

        accept 911 cost 0

      !

      voice grouped-trunk SIP_GROUP_2

        description "Test PBX"

        trunk T04

        accept 555999643X cost 0

       

       

       

       

      I can list a lot more debugs, but this is the first Rx and Tx group on a call from FreePBX that fails:

      192.168.1.21 is the Adtran IP.

      208.x.x.x is our office IP

      1234567890 is the cell phone I'm calling

      1234561234 is our caller ID / main number

       

       

      15:48:41.289 SIP.STACK MSG     Rx: UDP src=192.168.1.8:5060 dst=192.168.1.21:5060

      15:48:41.289 SIP.STACK MSG         INVITE sip:1234567890@192.168.1.21:5060 SIP/2.0

      15:48:41.289 SIP.STACK MSG         Via: SIP/2.0/UDP 208.x.x.x:5060;rport;branch=z9hG4bKPj2bcc7676-3f47-46b8-87f4-ef4266811                                                                                                                957

      15:48:41.290 SIP.STACK MSG         From: <sip:1234561234@192.168.1.8>;tag=2f8cc17b-ea32-46d5-805b-0288b366ef16

      15:48:41.290 SIP.STACK MSG         To: <sip:1234567890@192.168.1.21>

      15:48:41.290 SIP.STACK MSG         Contact: <sip:asterisk@208.x.x.x:5060>

      15:48:41.290 SIP.STACK MSG         Call-ID: b360c72f-3a73-4d15-8599-e87fdf557d3a

      15:48:41.290 SIP.STACK MSG         CSeq: 26705 INVITE

      15:48:41.290 SIP.STACK MSG         Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGIS                                                                                                                TER, REFER, MESSAGE

      15:48:41.291 SIP.STACK MSG         Supported: 100rel, timer, replaces, norefersub

      15:48:41.291 SIP.STACK MSG         Session-Expires: 1800

      15:48:41.291 SIP.STACK MSG         Min-SE: 90

      15:48:41.291 SIP.STACK MSG         Max-Forwards: 70

      15:48:41.291 SIP.STACK MSG         User-Agent: FPBX-13.0.192.9(13.14.0)

      15:48:41.291 SIP.STACK MSG         Content-Type: application/sdp

      15:48:41.292 SIP.STACK MSG         Content-Length:   312

      15:48:41.292 SIP.STACK MSG

      15:48:41.292 SIP.STACK MSG         v=0

      15:48:41.292 SIP.STACK MSG         o=- 430554945 430554945 IN IP4 192.168.1.8

      15:48:41.292 SIP.STACK MSG         s=Asterisk

      15:48:41.292 SIP.STACK MSG         c=IN IP4 208.x.x.x

      15:48:41.293 SIP.STACK MSG         t=0 0

      15:48:41.293 SIP.STACK MSG         m=audio 10578 RTP/AVP 0 8 3 111 101

      15:48:41.293 SIP.STACK MSG         a=rtpmap:0 PCMU/8000

      15:48:41.293 SIP.STACK MSG         a=rtpmap:8 PCMA/8000

      15:48:41.293 SIP.STACK MSG         a=rtpmap:3 GSM/8000

      15:48:41.293 SIP.STACK MSG         a=rtpmap:111 G726-32/8000

      15:48:41.294 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000

      15:48:41.294 SIP.STACK MSG         a=fmtp:101 0-16

      15:48:41.294 SIP.STACK MSG         a=ptime:20

      15:48:41.294 SIP.STACK MSG         a=maxptime:150

      15:48:41.294 SIP.STACK MSG         a=sendrecv

      15:48:41.295 SIP.STACK MSG

      15:48:41.299 SIP.STACK MSG     Tx: UDP src=192.168.1.21:5060 dst=192.168.1.8:5060

      15:48:41.300 SIP.STACK MSG         SIP/2.0 100 Trying

      15:48:41.300 SIP.STACK MSG         From: <sip:1234561234@192.168.1.8>;tag=2f8cc17b-ea32-46d5-805b-0288b366ef16

      15:48:41.300 SIP.STACK MSG         To: <sip:1234567890@192.168.1.21>

      15:48:41.300 SIP.STACK MSG         Call-ID: b360c72f-3a73-4d15-8599-e87fdf557d3a

      15:48:41.300 SIP.STACK MSG         CSeq: 26705 INVITE

      15:48:41.301 SIP.STACK MSG         Via: SIP/2.0/UDP 208.x.x.x:5060;received=192.168.1.8;rport=5060;branch=z9hG4bKPj2bcc767                                                                                                                6-3f47-46b8-87f4-ef4266811957

      15:48:41.301 SIP.STACK MSG         Contact: <sip:1234567890@192.168.1.21:5060;transport=UDP>

      15:48:41.301 SIP.STACK MSG         Supported: 100rel,replaces

      15:48:41.301 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

      15:48:41.301 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908e_2nd_Gen/R10.3.3.E

      15:48:41.301 SIP.STACK MSG         Content-Length: 0

       

       

      I'm really confused why the Contact is our office IP with FreePBX, and the local IP for our production PBX.  Maybe I have the firewall setup incorrectly on FreePBX?  Besides that, do the trunks, groups, and accept lines look like they will work?

       

       

      Here are the first few lines from a successful call from our production PBX:

      15:54:57.235 SIP.STACK MSG     Rx: UDP src=192.168.1.10:5060 dst=192.168.1.21:5060

      15:54:57.235 SIP.STACK MSG         INVITE sip:1234567890@192.168.1.21:5060 SIP/2.0

      15:54:57.235 SIP.STACK MSG         To: "City State" <sip:1234561234@192.168.1.21:5060>

      15:54:57.236 SIP.STACK MSG         From: <sip:1234561234@PBX.fqdn.com:5060>;tag=3hIbAzy

      15:54:57.236 SIP.STACK MSG         Via: SIP/2.0/UDP 10.100.124.10;branch=z9hG4bKlfnpvth2EbzfmAOytcgj

      15:54:57.236 SIP.STACK MSG         Call-ID: c41aaf0dc6279f489e53fbd738e77850@192.168.1.10

      15:54:57.236 SIP.STACK MSG         CSeq: 1 INVITE

      15:54:57.236 SIP.STACK MSG         Contact: <sip:1234561234@192.168.1.10>

      15:54:57.236 SIP.STACK MSG         Max-Forwards: 70